A method of smart coding mode switching includes receiving a first data including a primary copy and a partial copy. The method includes determining if switching a coding mode from channel aware mode to non-channel aware mode may be advantageous. The method further includes transmitting a request to another device for coding mode switch in response to determination result. The method includes receiving and decoding of a second data that includes a primary copy.

An apparatus for adapting input audio information, encoding one or more audio objects, to obtain adapted audio information is provided. The input audio information includes two or more input audio downmix channels and further includes input parametric side information. The adapted audio information includes one or more adapted audio downmix channels and further includes adapted parametric side information. The apparatus includes a downmix signal modifier for adapting, depending on adaptation information, the two or more input audio downmix channels to obtain the one or more adapted audio downmix channels. Moreover, the apparatus includes a parametric side information adapter for adapting, depending on the adaptation information, the input parametric side information to obtain the adapted parametric side information.

A system having a head-mounted display (HMD) mount and a mobile device, are described. A processor of the system can determine whether the mobile device is mounted on the HMD mount and handle an audio signal communicated from the mobile device to a wireless headphone based on whether the mobile device is mounted on the HMD mount. When the mobile device is not mounted on the HMD mount, the mobile device or the wireless headphone may operate in a first audio mode. When the mobile device is mounted on the HMD mount, the mobile device or the wireless headphone may operate in a second audio mode. The second audio mode can reduce audio signal latency between the mobile device and the wireless headphone and increase motion-to-sound quality. Other embodiments are also described and claimed.

A method of smart coding mode switching includes receiving a first data including a primary copy and a partial copy. The method includes determining if switching a coding mode from channel aware mode to non-channel aware mode may be advantageous. The method further includes transmitting a request to another device for coding mode switch in response to determination result. The method includes receiving and decoding of a second data that includes a primary copy.

An apparatus including means configured to: obtain at least one audio stream, wherein the at least one audio stream includes one or more transport audio signals, wherein the one or more transport audio signals is a defined type of transport audio signal; and convert the one or more transport audio signals to at least one or more further transport audio signals, the one or more further transport audio signals being a further defined type of transport audio signal.

Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.

Systems and methods are described for concealing packet loss in a received audio stream. Packets of the audio stream may be received in a non-lapped transform domain format, where at least one packet is missing in the stream. The received packets are decoded, and each missing packet in the decoded stream is replaced by a reduced-energy signal block. Each reduced-energy signal block may also be modified at a beginning or ending boundary, and shifted such that a start or end of each missing packet does not coincide with a peak of a transform window of a lapped transform domain format. The raw audio signal may then be encoded into transform windows having the lapped transform domain format. Packet loss concealment may then be performed for selected transform windows that include modified reduced-energy blocks, either prior to transmission or after transmission by the receiving endpoint.

Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.

A method of providing audio information, said method comprising: (i) receiving audio filter settings in a client device; (ii) receiving audio data in the client device, wherein the received audio data is in an audio coding format; (iii) converting the audio filter settings to an audio filter signal in a processor of the client device, where the audio filter signal is a time-varying signal; (iv) converting the received audio data to an audio signal in a processor of the client device, where the audio signal is a time-varying signal; and (v) transmitting the converted audio filter signal and the converted audio signal from the client device to an audio output device, where the audio output device is separate from and in communication with the client device, and where the audio output device is configured for modifying the audio signal according to the audio filter signal to generate a time-varying audio output.

An apparatus for downmixing three or more audio input channels to obtain two or more audio output channels is provided. The apparatus includes a receiving interface for receiving the three or more audio input channels and for receiving side information. Moreover, the apparatus includes a downmixer for downmixing the three or more audio input channels depending on the side information to obtain the two or more audio output channels. The number of the audio output channels is smaller than the number of the audio input channels. The side information indicates a characteristic of at least one of the three or more audio input channels, or a characteristic of one or more sound waves recorded within the one or more audio input channels, or a characteristic of one or more sound sources which emitted one or more sound waves recorded within the one or more audio input channels.